Sigapy: a failed attempt to create a product in Paraguay

A little more than a year ago, I created a service which main intention was to provide my fellow Paraguayans living abroad, an easy way to call home using their smartphones. It was not like Viber or Tango app, you could actually call to land lines and mobile phones, and of course between Sigapy users.

There’s no specialized service like this in Paraguay, and I wanted to fill that void with something new and easy to use, and so Sigapy was born. It was conceived as an academical project but soon it started to appear exploitable as a commercial service, but the environment for this kind of business in Paraguay isn’t the best (actually, it’s horrible) and I had to shut it down for my own juridical sake.

Today, I decided someone else can give it a try, because it’s a more or less mature product and can easily be adapted to someone else’s needs.

Check out the videos and the landing page I created. The videos are in Spanish, but the flow can be understood by anyone.

Click to Call application using webrtc2sip + asterisk

I’ve been working on this for the last few days and I pleased to say that I managed to get through the series of problems that the learning curve entails and now the app is finally working.

My intention was to learn the fundamentals of webRTC and SIP over websockets and I haven’t found a better solution than the one offered by Doubango Telecom. Their impressive job is transforming the way we communicate and I want to be part of that transformation when it finally hit the “standard technology” label.

The purpose of the application is to demonstrate a new way of doing a click to call service. There are a variety of similar solutions with different approaches being the Java applet the most commonly used amongst them.

Well, although this app doesn’t bring any new thing into the world, it certainly serves the purpose of demonstrating a new way of making things. It’s entirely made using HTML5 with the javascript library that made SIPML5 possible.

The app connects to my lab’s Asterisk, via webrtc2sip which deals with the SIP over WS on one side, and SIP over UDP in the other part. The media is also handled by webrtc2sip by translating the SDP profiles and making transcoding on demand.

You can try it here. It is necessary the latest version of Chrome stable or Firefox Nightly. If you don’t meet the minimum needs, the page will just stay there doing nothing. In case you get an error message it’s probably because my server crashed. Please let me know if that happen.

You can browse the javascript code to see how simple it is. I’m planning to turn the project into something more elaborated and then publish the code. By now, it’s too simple to bother 😉


For some reason, behind very restrictive firewalls the audio is not working properly. I’m working on this to fix it ASAP. Please let me know if that’s your case.


I temporarily deactivated the demo because my server suffered large amounts of hacking attempts. I knew this was possible but I wasn’t expecting this volume.